Restful HTTP Webrtc Testing

RESTful HTTP WebRTCRESTful HTTP Testing Real-time voice and video quality of service (QoS)for allstreams SRTP and SRTCP multiplexing tests andHTTP HTTPs support 1.28 million SRTP and SRTCP streams 1.28 million TLS and DTLS sessions 1.28 million WebRTC endpoints FEATURESKEY HTTP network and service testing throughout development and deployment. Easy-to-use, high-performance and high-capacity load testing solution to enable OMA RESTfulWebRTC-compliant device, F LEX TEST APPLICATION TEST Mix of VoIP, VoLTE and WebRTC service tests IMS/VoLTE and VoIP core network emulation Policy and charging function (PCRF)emulation Both ICE-Lite and ICE-Full support STUN, TURNand ICENAT server tests OPUS and VP8WebRTC codecs

SPEC SHEET RESTful HTTP WebRTC Testing

OVERVIEW Web real-time communications (WebRTC) technology enables real-time communication via browsers. Service providers have deployed WebRTC and are doing trials with the technology to extend their IMS services such as voice over IP (VoIP), voice over LTE (VoLTE), video over LTE (ViLTE) and rich communication services (RCS) to Internet (web) users, hence enabling real-time communication between IMS and web users such as voice, video and chat. The interworking between IMS and web users is made possible by the WebRTC gateway, which translates signaling and transcodes media between these two technology domains.

TELECOM WEB

SIP Web Signaling RTP/SRTP DTLS/SRTP

WebRTC Gateway

Figure 1. Interworking via WebRTC gateway

WebRTC GATEWAY A WebRTC gateway is typically implemented and deployed in one of the two following variations: 1. Session border controller (SBC) supporting WebRTC signaling (eP-CSCF), media (e-IMS-AGW) and NAT (STUN/TURN/ICE) functions.

STUN/TURN/ICE NAT

Web Signaling SIP eP-CSCF

DTLS-SRTP RTP eIMS-AGW

SBC

2. SBC supporting only WebRTC media (eIMS-AGW) function, and a separate device supporting WebRTC signaling (WebRTC signaling gateway) and STUN/TURN/ICE (NAT) functions.

STUN/TURN/ICE NAT

Web Signaling WebRTC SIP SIP Sig GW P-CSCF

DTLS-SRTP RTP eIMS-AGW

SBC RESTful HTTP WebRTC Testing

RESTFUL HTTP WebRTC specifications do not define standard web signaling protocol, leaving web application architects and developers to decide. The signaling protocol is a requirement to transmit and exchange user device capabilities such as codec and ICE credentials from one endpoint to another in order to set up the endpoint for a communication session. Despite the flexibility and freedom provided to the application designers to choose the signaling protocol, having no standard signaling protocol introduces challenges in delivering interoperable WebRTC services. To overcome the interoperatbility challenge, Open Mobile Alliance (OMA) has defined a standardized RESTful HTTP network API that allows a a web-user application (e.g., Javascript running in a WebRTC-enabled browser) to signal video, voice and data session over IP with another communication endpoint in the network. It includes common data types, naming conventions, fault definitions and namespaces.

HTTPFLEX APPLICATION The httpFlex application emulates RESTful HTTP WebRTC enpoints in compliance with OMA RESTful Network API for WebRTC signaling v1.0 specifications. Its graphical user interface is flexible and easy to use, enabling customers to use out-of-box pre-canned call flows or to define their own custom call flows. It can be used to simulate real-world normal and abnormal calls during various lifecycle phases from lab testing to live network testing. The httpFlex application runs over the QualityAssurer QA-805 test platform.

TEST CONFIGURATION The QualityAssurer QA-805 test platform–the industry’s leading high-performance and high-capacity platform–supports the comprehensive set of applications and features to test the WebRTC gateway as well as the network end to end. The following picture depicts both test configurations: 1) SBC supporting WebRTC signaling (eP-CSCF), media (e-IMS-AGW) and NAT (STUN/TURN/ICE) functions. In this test configuration, the httpFlex application is used together with EXFO’s volteFlex, proxyFlex, and hssFlex applications for SBC wraparound testing; httpFlex emulates WebRTC endpoints, volteFlex emulates VoIP/VoLTE/ViLTE endpoints, proxyFlex emulates CSCFs in the IMS core and hssFlex emulates the HSS and PCRF.

SUT = SBC

STUN/TURN/ICE NAT

EXFO 4G RESTful HTTP SIP SIP eP-CSCF IMS CORE

EXFO EXFO DTLS-SRTP (OPUS/VP8) RTP (G.711, H.264) WebRTC UE VoLTE UE eIMS-AGW

Diameter

EXFO

PCRF RESTful HTTP WebRTC Testing

2) End-to-end IMS network testing with a mix of different types of endpoints, such as WebRTC, VoLTE and ViLTE endpoint. In this test configuration, httpFlex is used together with the volteFlex and hssFlex applications to test the IMS network; httpFlex emulates WebRTC endpoints, volteFlex emulates VoIP/VoLTE/ViLTE endpoints while hssFlex emulates PCRF.

SUT = IMS

STUN/TURN/ICE HSS AS

4G RESTful HTTP SIP PCRF CSCF

EXFO EXFO RTP (G.711, H.264) WebRTC UE DTLS-SRTP (OPUS/VP8) VoLTE UE SBC

Diameter

EXFO

PCRF

KEY USE CASES

Simulates millions of WebRTC enpoints and other network devices for WebRTC gateway testing, thus enabling selection of the right gateway for WebRTC deployment

Tests that the WebRTC gateway is able to scale linearly to accomadate growing loads without any performance/capacity degradation

Tests how many RESTful HTTP sessions the WebRTC gateway is able to set up per second, hold and maintain successfully over a period of time with different notification refresh timeout values

Verifies the impact on WebRTC gateway performance as a result of resource-intensive ICE procedures that include prioritization of IP candidates and pair exchange, and order them into a checklist

Validates how many concurrent signaling and media sessions WebRTC gateway (data transport layer security-secure real-time transport protocol, DTLS-SRTP/SRTCP, and DTLS-SRTP/SRTCP multiplexing) can support concurrently

Verifies resource-intensive session-traversal-utilities-for-NAT (STUN) and traversal-using-relays-around-NAT (TURN) authentication using short-term ICE attributes (ice-ufrag and ice-pwd) credentials and DTLS fingerprint authentication in order to verify that the certificate presented in the DTLS handshake does not deteriorate the WebRTC gateway performance

Benchmarks the maximum number of concurrent DTLS-SRTP voice and video media sessions that the WebRTC gateway is able to set up and maintain simultaneously, both with and without transcoding

Generates and analyzes line rate (1G and 10G) voice and video streams with a mix of codecs, such as OPUS, VP8, H.264 and G.711

Checks that the WebRTC gateway is able to deliver expected QoS with different quality-parameter settings, such as ToS/differentiated services code point (DSCP), VLAN and MPLS

Performs high-availability tests to determine resilience of the WebRTC gateway under overload conditions, security attacks, card and port failover scenarios

Exercises the entire operator network infrastructure by testing end-to-end service delivery with a mix of WebRTC, VoIP, VoLTE, ViLTE and RCS/message session relay protocol (MSRP) services RESTful HTTP WebRTC Testing

SPECIFICATIONS Platform QA-805

W2CM-10GbE (8 X 1 GigE and 2 X 10 GigE) W2CM-10GbE-Lite (8 X 1 GigE and 2 X 10 GigE) Modules and interfaces W2CM-4GbE (4 X 1 GigE) W2CM-Sig (8 X 1 GigE and 2 X 10 GigE, signaling only)

WebRTC protocols RESTful HTTP, ICE-Full/Lite, STUN, TURN, DTLS-SRTP, SRTP and SRTCP multiplexing

Transport and IP protocols TCP, TLS, IPv4, IPv6

WebRTC codecs OPUS, G.711, VP8 and H.264

RESTful HTTP WebRTC endpoint capacity 1.28 million per QA-805 platform

Number of DTLS-SRTP sessions 1.28 million concurrent DTLS-SRTP and DTLS-SRTCP streams per QA-805 platform

Voice (ITU-T G.107 E-Model) Quality measurements Video (RFC 4445 – VQT MDI) Jitter, loss, delay, etc.

Network configuration Unique MAC addresses, VLAN tag, MPLS label, ToS and DSCP settings

IP (IPv4 to IPv6) Transport (TCP to TLS to TCP/UDP/SCTP) Interworking Signaling (RESTful HTTP to SIP) Media (RTP/RTCP to SDES-SRTP/SRTCP to DTLS-SRTP/SRTCP) Note: For interworking testing volteFlex application is required

Signaling trace monitor, call records, user-defined KPI, summary and call-flow statistics, table, histogram Statistics and logging and chart format, and report generation in html and .csv

Create invalid WebRTC messages, create invalid and error call flow, mix valid and invalid call, Negative testing STUN/TURN inactivity and connectivity failure, etc.

Automation TCL command line interface

ORDERING INFORMATION

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